Fix issue with conflicting behavior if setting a max BW with b=AS on both audio and video.

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 244 matching lines @@
   int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   int dtmf_payload_freq_ = -1;
   bool recv_transport_cc_enabled_ = false;
   bool recv_nack_enabled_ = false;
   bool desired_playout_ = false;
   bool playout_ = false;
   bool send_ = false;
   webrtc::Call* const call_ = nullptr;
-  webrtc::Call::Config::BitrateConfig bitrate_config_;
 
   // Queue of unsignaled SSRCs; oldest at the beginning.
   std::vector<uint32_t> unsignaled_recv_ssrcs_;
 
   // Volume for unsignaled streams, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
   // Sink for latest unsignaled stream - may be set before the stream exists.
   std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
   // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
   // and https://code.google.com/p/chromium/issues/detail?id=547661
   uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_